Re: [asterisk-users] video mail is not store
One thing i have noticed is that your profile-id don't match and therefore you would get no video. Asterisk is not a problem On Jan 9, 2012, at 1:20 AM, Durgesh Mishra wrote: Hi, I am facing an issue while testing the video mail service of Asterisk. I have two different setup on one setup client being used is Mercuro while on the other client is Android based. On the Mercuro setup video mail is stored and retrieved properly while with Android based setup video mail is not stored (audio is through). Both the client use H.264 codec with following sdp information: Android Based Client SDP Parameters v=0 o=- 1325786904 1325786904 IN IP4 172.16.130.47 s=Polycom RealPresence c=IN IP4 172.16.130.47 b=AS:1920 t=0 0 a=sendrecv m=audio 3230 RTP/AVP 118 115 114 113 0 8 119 a=rtpmap:118 SIRENLPR/48000 a=fmtp:118 bitrate=64000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:114 G7221/32000 a=fmtp:114 bitrate=32000 a=rtpmap:113 G7221/32000 a=fmtp:113 bitrate=24000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:119 telephone-event/8000 a=fmtp:119 0-15 m=video 3232 RTP/AVP 109 110 a=rtcp-fb:* ccm fir tmmbr a=rtpmap:109 H264/9 a=fmtp:109 profile-level-id=42800d; max-mbps=108000; max-fs=3840; max-br=1920; sar=13 a=rtpmap:110 H264/9 a=fmtp:110 profile-level-id=42800d; packetization-mode=1; max-mbps=108000; max-fs=3840; max-br=1920; sar=13 m=application 3236 RTP/AVP 100 a=sendrecv a=rtpmap:100 H224/4800 MERCURO SDP Parameters v=0 o=- 1234 1235 IN IP4 10.34.77.90 s=Mercuro IMS Client Session t=0 0 m=audio 31098 RTP/AVP 0 8 101 c=IN IP4 10.34.77.90 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=silenceSupp:off - - - - a=sendrecv m=video 34442 RTP/AVP 113 c=IN IP4 10.34.77.90 a=rtpmap:113 H264/9 a=fmtp:113 fmtp:113 profile-level-id=42e00a; packetization-mode=1; max-br=2000; max-mbps=11880 a=sendrecv Plz tell me is there any limitation from the Asterisk side i.e. H.264 codec is supported only with limited parameters. I would like to know what parameters of H.264 codec are supported by Asterisk? Your comnments are most welcome. Regards, Durgesh. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk refuses INVITE (401) and I don't know why
just a quick observation, but not sure that it is critical in this case, the first invite comes without Authorization header, then gets challenged then resends the invite (with increased cseq) with calculated response based on the challenge from the server. In your AAstra case, the first invite already contained Authorization header (which is really impossible because you don't have all the pieces to calculate the response). Normally not an issue, as UAS should challenge it, but I wonder why it does it anyway. I would compare Authorize elements between 2 cases particularly response, uri and authorization user name. if response is the same between the two, I am lost. On Nov 24, 2011, at 2:11 PM, Jonas Kellens wrote: On 11/22/2011 06:13 PM, Alex Vishnev wrote: it is strange that Aastra acks 401, sends another invite but does not increase CSeq. Is that the same behavior with others? On Nov 22, 2011, at 11:51 AM, Jonas Kellens wrote: This is a trace taken when an Alcatel-Lucent PBX sends an INVITE (no refusal by Asterisk). Do you see any difference ? A1.A1.A1.A1 = IP-address Asterisk PBX AL.AL.AL.AL = IP-address Alcatel-Lucent PBX --- SIP read from UDP:AL.AL.AL.AL:5060 --- INVITE sip:311083335533@A1.A1.A1.A1;user=phone SIP/2.0 Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, SUBSCRIBE, OPTIONS, UPDATE Supported: replaces, timer, 100rel User-Agent: OmniPCX Enterprise R9.1 i1.605.21 Session-Expires: 1800;refresher=uac Min-SE: 900 P-Asserted-Identity: Dan Luc sip:328883300@192.168.8.10;user=phone To: sip:311083335533@A1.A1.A1.A1;user=phone From: Dan Luc sip:328883...@al.al.al.al:5060;user=phone;tag=37a49f0486bab42b240be214b2d13153 Contact: sip:328883...@al.al.al.al:5060;transport=UDP Call-ID: 2fae0b0266919172cac1e23dc2567cd2@192.168.8.10 CSeq: 443337258 INVITE Via: SIP/2.0/UDP AL.AL.AL.AL:5060;branch=z9hG4bK5dee58e3294e4b7f9fe34c65af7b4cae Max-Forwards: 70 Content-Type: application/sdp Content-Length: 292 v=0 o=OXE 1322045354 1322045354 IN IP4 AL.AL.AL.AL s=abs c=IN IP4 AL.AL.AL.AL t=0 0 m=audio 34422 RTP/AVP 8 18 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=ptime:20 a=maxptime:40 a=rtpmap:97 telephone-event/8000 --- Reliably Transmitting (NAT) to AL.AL.AL.AL:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP AL.AL.AL.AL:5060;branch=z9hG4bK5dee58e3294e4b7f9fe34c65af7b4cae;received=AL.AL.AL.AL From: Dan Luc sip:328883...@al.al.al.al:5060;user=phone;tag=37a49f0486bab42b240be214b2d13153 To: sip:311083335533@A1.A1.A1.A1;user=phone;tag=as1b6f387a Call-ID: 2fae0b0266919172cac1e23dc2567cd2@192.168.8.10 CSeq: 443337258 INVITE Server: Asterisk PBX 1.6.2.20 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=domain.tld, nonce=7684ab1d Content-Length: 0 --- SIP read from UDP:AL.AL.AL.AL:5060 --- INVITE sip:311083335533@A1.A1.A1.A1;user=phone SIP/2.0 Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, SUBSCRIBE, OPTIONS, UPDATE Supported: replaces, timer, 100rel User-Agent: OmniPCX Enterprise R9.1 i1.605.21 Session-Expires: 1800;refresher=uac Min-SE: 900 P-Asserted-Identity: Dan Luc sip:328883300@192.168.8.10;user=phone To: sip:311083335533@A1.A1.A1.A1;user=phone From: Dan Luc sip:328883...@al.al.al.al:5060;user=phone;tag=37a49f0486bab42b240be214b2d13153 Contact: sip:328883...@al.al.al.al:5060;transport=UDP Call-ID: 2fae0b0266919172cac1e23dc2567cd2@192.168.8.10 CSeq: 443337259 INVITE Max-Forwards: 70 Authorization: Digest username=SIPPEERusername,realm=domain.tld,nonce=7684ab1d,algorithm=MD5,uri=sip:311083335533@A1.A1.A1.A1;user=phone,response=38bb824b9081bf2eefe9f9677d3eb005 Via: SIP/2.0/UDP AL.AL.AL.AL:5060;branch=z9hG4bK52dae2e7816406e20a9c02aa9cb86726 Content-Type: application/sdp Content-Length: 292 v=0 o=OXE 1322045354 1322045354 IN IP4 AL.AL.AL.AL s=abs c=IN IP4 AL.AL.AL.AL t=0 0 m=audio 34422 RTP/AVP 8 18 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=ptime:20 a=maxptime:40 a=rtpmap:97 telephone-event/8000 --- Transmitting (NAT) to AL.AL.AL.AL:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP AL.AL.AL.AL:5060;branch=z9hG4bK52dae2e7816406e20a9c02aa9cb86726;received=AL.AL.AL.AL From: Dan Luc sip:328883...@al.al.al.al:5060;user=phone;tag=37a49f0486bab42b240be214b2d13153 To: sip:311083335533@A1.A1.A1.A1;user=phone Call-ID: 2fae0b0266919172cac1e23dc2567cd2@192.168.8.10 CSeq: 443337259 INVITE Server: Asterisk PBX 1.6.2.20 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Session-Expires: 1800;refresher=uac Contact: sip:311083335533@A1.A1.A1.A1 Content-Length: 0 Thanks ! Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com
Re: [asterisk-users] Asterisk refuses INVITE (401) and I don't know why
do you see the register messages? if your device is not registered, INVITE would be challenged. You should check to see if register messages are being properly acknowledge with 200OK. On Nov 22, 2011, at 10:29 AM, Jonas Kellens wrote: On 11/22/2011 04:25 PM, Bruce Ferrell wrote: Jonas, May I suggest that you present us your sip.conf entry for this peer, properly redacted, of course. That might help more. What I do for gateways at known addresses is to put an entry like this into the sip.conf entry: [peer] type=peer defaultip=192.168.40.123 insecure=invite,port context=some_context This is the peer definition in sip.conf : [SIPPEERusername] type=friend host=dynamic defaultuser=SIPPEERusername secret=guessthis context=from-PEERTRUNK nat=yes dtmfmode=rfc2833 canreinvite=no disallow=all allow=alaw allow=gsm Hope you can help me out with this extra information. Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk refuses INVITE (401) and I don't know why
Your registration should have also have the flow PEER ASTERISK REGISTER--- --401 REGISTER(nonce) - 200OK Then the server controls the life of the registration and 200 Expires Header gives you this timeout. If the invite is sent within that window, then Asterisk should not challenge anymore. If Invite is challenged and the peer responds with the correctly calculated NONCE, domain and other Auth params, then something is wrong with your Authentication. I suggest trapping the traffic with Ethereal or any other packet capture programs and examining that carefully from the start of the session (i.e. register) to the invite. I would also check where the 401 is coming from (i.e. IP address). Hope that helps Alex On Nov 22, 2011, at 11:23 AM, Jonas Kellens wrote: On 11/22/2011 04:37 PM, Bruce Ferrell wrote: On 11/22/2011 07:29 AM, Jonas Kellens wrote: On 11/22/2011 04:25 PM, Bruce Ferrell wrote: Jonas, May I suggest that you present us your sip.conf entry for this peer, properly redacted, of course. That might help more. What I do for gateways at known addresses is to put an entry like this into the sip.conf entry: [peer] type=peer defaultip=192.168.40.123 insecure=invite,port context=some_context This is the peer definition in sip.conf : [SIPPEERusername] type=friend host=dynamic defaultuser=SIPPEERusername secret=guessthis context=from-PEERTRUNK nat=yes dtmfmode=rfc2833 canreinvite=no disallow=all allow=alaw allow=gsm Hope you can help me out with this extra information. Kind regards, Jonas. From what I see in your entry, you are requiring registration from the peer. The next thing i would check is to see if the registration has succeeded. If it doesn't succeed, you will see the results you presented. I see you have the peer set as a dynamic host, and if the IP address of the device does in fact change then registration is appropriate. Registration of the SIP PEER is no problem. The PEER registers with a correct REGISTER statement and Asterisk sends a 200 OK. So the PEER is registered and then wants to make a call (INVITE) but for some reason this INVITE is being refused with 401-Unauthorized. The first 401-Unauthorized is normal, because the SIP PEER needs to send a second INVITE with a challenge (nonce). But after this INVITE with challenge, Asterisk still sends a 401 and that's strange !! Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk refuses INVITE (401) and I don't know why
I doubt it. Unknown headers should be ignored by UAS. is it possible to post the trace? On Nov 22, 2011, at 11:39 AM, Jonas Kellens wrote: On 11/22/2011 05:31 PM, Alex Vishnev wrote: Your registration should have also have the flow PEER ASTERISK REGISTER--- --401 REGISTER(nonce) - 200OK Then the server controls the life of the registration and 200 Expires Header gives you this timeout. If the invite is sent within that window, then Asterisk should not challenge anymore. If Invite is challenged and the peer responds with the correctly calculated NONCE, domain and other Auth params, then something is wrong with your Authentication. I suggest trapping the traffic with Ethereal or any other packet capture programs and examining that carefully from the start of the session (i.e. register) to the invite. I would also check where the 401 is coming from (i.e. IP address). Hope that helps Alex I've already captured with Wireshark, but what to do with it if I don't know what I'm looking for ?? Registration goes without problem, but every INVITE is answered with a 401-Unauthorized. Like I already said : there is no problem with Avaya, Panasonic and Alcatel-Lucent. The only difference I see between an INVITE from Avaya and the INVITE from Aastra PBX is the presence of the SIP-header : P-Behind-Gsi: 192.168.6.1. Could this header mess up Asterisk ? Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk refuses INVITE (401) and I don't know why
it is strange that Aastra acks 401, sends another invite but does not increase CSeq. Is that the same behavior with others? On Nov 22, 2011, at 11:51 AM, Jonas Kellens wrote: On 11/22/2011 05:42 PM, Alex Vishnev wrote: I doubt it. Unknown headers should be ignored by UAS. is it possible to post the trace? On Nov 22, 2011, at 11:39 AM, Jonas Kellens wrote: What trace do you need ? Have you read my original post ? Asterisk SIP debug trace is posted in my original post. Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Security: Allow only one phone per sip registration
the best way to handle large sip client base is using provisioning interface. Even though you can create configuration files and server them with asterisk+extensions, you need to consider security aspects of this approach as well. Using tftp or simple protocols to server config files works on LAN, but does not scale for large installs (my opinion). HTTP is a better choice, but then all the information is passed in clear. HTTPS is obviously a better choice with SSL, but if your devices can't handle SSL it will become a problem. A good solution is to provide a mix depending on your SIP client capabilities. In the configuration you can supply password/secret as other recommend and any other device specific configuration (i.e. preferred codec, DNS, etc). it really becomes a powerful tool. You also need to have a management capabilities to generate and update your configuration profile either for individual devices (i.e. changes users's secret) or in bulk (change DNS servers or proxy on 1000 SIP clients at once). SIP clients will also need to have capabilities to poll for this configuration on reboot or on regular poll intervals. If you are doing that on the poll interval, don't make it the interval too short (i.e. minutes). I would say 3-4 times a day is a good starting point. If your network is pretty static and not much information changes you can even make it 1-2 a day and experiment with your network load. On Oct 14, 2011, at 7:09 AM, A J Stiles wrote: On Friday 14 October 2011, Muro, Sam wrote: Hi there Consider this. You have three SIP extension 200, 201 and 202 and you have configured your phones, say Polycom 331 to those accounts. 200 being one very sensitive individual. Lets say, an insider, get a new phone or perhaps an xlite and configure it with the same extension, 200. Asterisk will register it as 200 to the new IP address. Now extension 202 call 200. The hacker answers it and pretend is the same person. Do what he want to do and thats it. Question; How can i stop this type of threat Be careful who you employ and how you treat them :) Once someone has physical access to your equipment, all bets are off . -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP OPTIONS... Error?
no, you need a tag i.e From: sip:p...@xx.xx.xx.xx;tag=xxx, where xx is a unique identifier see the definition of SIP Dialog Dialog: A dialog is a peer-to-peer SIP relationship between two UAs that persists for some time. A dialog is established by SIP messages, such as a 2xx response to an INVITE request. A dialog is identified by a call identifier, local tag, and a remote tag. A dialog was formerly known as a call leg in RFC 2543. On Sep 19, 2011, at 1:11 PM, Bruce Ferrell wrote: On 09/19/2011 09:33 AM, Alex Balashov wrote: Every request needs a From tag. Uh... OK. Isn't this a From tag: From: sip:p...@xx.xx.xx.xx Line three of what I send? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Office 365 Unified Messaging... anyone done it?
this could be an unsupported codec. Do you know if Office365 supports PCMU? I would also try to get rid of 101 (rfc2833) and see if that makes a difference On Aug 15, 2011, at 8:40 PM, o o wrote: Trying to make this work, and Office 365 support is useless, giving me the following response when I asked them for help troubleshooting a 488 Not Acceptable Here. Regarding your service request about configuring your PBX system with Office 365, we do not support specific setups for PBX systems for Unified Messaging. Please contact the vendor for more specific instructions and configurations. Here is a SIP debug: [2011-08-11 23:00:26] VERBOSE[17000] chan_sip.c: Reliably Transmitting (no NAT) to 65.55.174.100:5061: OPTIONS sip:um.outlook.com SIP/2.0 Via: SIP/2.0/TLS 1.2.3.4:5061;branch=z9hG4bK16884162 Max-Forwards: 70 From: Unknown sip:Unknown@1.2.3.4;tag=as438c582c To: sip:um.outlook.com Contact: sip:Unknown@1.2.3.4:5061;transport=TLS Call-ID: 67f260947dae7c27121ca30e5ee9d3ef@1.2.3.4:5061 CSeq: 102 OPTIONS User-Agent: FPBX-2.8.1(1.8.5.0) Date: Fri, 12 Aug 2011 06:00:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2011-08-11 23:00:27] VERBOSE[17375] chan_sip.c: --- SIP read from TLS:65.55.174.100:5061 --- SIP/2.0 200 OK Via: SIP/2.0/TLS 1.2.3.4:5061;branch=z9hG4bK16884162 From: Unknown sip:Unknown@1.2.3.4;tag=as438c582c To: sip:um.outlook.com;tag=b4ec76231 Call-ID: 67f260947dae7c27121ca30e5ee9d3ef@1.2.3.4:5061 CSeq: 102 OPTIONS ACCEPT: application/sdp CONTENT-LENGTH: 0 ALLOW: INVITE ALLOW: BYE ALLOW: CANCEL ALLOW: OPTIONS ALLOW: ACK ALLOW: INFO ALLOW: NOTIFY SERVER: RTCC/3.5.0.0 - [2011-08-11 23:00:27] VERBOSE[17375] chan_sip.c: --- (16 headers 0 lines) --- [2011-08-11 23:00:27] VERBOSE[17000] chan_sip.c: Really destroying SIP dialog '67f260947dae7c27121ca30e5ee9d3ef@1.2.3.4:5061' Method: OPTIONS [2011-08-11 23:00:47] VERBOSE[17999] chan_sip.c: Audio is at 5061 [2011-08-11 23:00:47] VERBOSE[17999] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [2011-08-11 23:00:47] VERBOSE[17999] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [2011-08-11 23:00:47] VERBOSE[17999] chan_sip.c: Reliably Transmitting (no NAT) to 65.55.174.100:5061: INVITE sip:9...@um.outlook.com SIP/2.0 Via: SIP/2.0/TLS 1.2.3.4:5061;branch=z9hG4bK70149e02 Max-Forwards: 70 From: Test User sip:210@1.2.3.4;tag=as746bc17a To: sip:9...@um.outlook.com Contact: sip:210@1.2.3.4:5061;transport=TLS Call-ID: 535c9a2d211f19be5528b2ce7dbce0d4@1.2.3.4:5061 CSeq: 102 INVITE User-Agent: FPBX-2.8.1(1.8.5.0) Date: Fri, 12 Aug 2011 06:00:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 238 v=0 o=root 1381221379 1381221379 IN IP4 1.2.3.4 s=Asterisk PBX 1.8.5.0 c=IN IP4 1.2.3.4 t=0 0 m=audio 17688 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [2011-08-11 23:00:47] VERBOSE[17375] chan_sip.c: --- SIP read from TLS:65.55.174.100:5061 --- SIP/2.0 100 Trying Via: SIP/2.0/TLS 1.2.3.4:5061;branch=z9hG4bK70149e02 From: Test User sip:210@1.2.3.4;tag=as746bc17a To: sip:9...@um.outlook.com Call-ID: 535c9a2d211f19be5528b2ce7dbce0d4@1.2.3.4:5061 CSeq: 102 INVITE Content-Length: 0 - [2011-08-11 23:00:47] VERBOSE[17375] chan_sip.c: --- (7 headers 0 lines) --- [2011-08-11 23:00:47] VERBOSE[17375] chan_sip.c: --- SIP read from TLS:65.55.174.100:5061 --- SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/TLS 1.2.3.4:5061;branch=z9hG4bK70149e02 From: Test User sip:210@1.2.3.4;tag=as746bc17a To: sip:9...@um.outlook.com;tag=aprqngfrt-hm4td72c6 Call-ID: 535c9a2d211f19be5528b2ce7dbce0d4@1.2.3.4:5061 CSeq: 102 INVITE Content-Length: 0 - [2011-08-11 23:00:47] VERBOSE[17375] chan_sip.c: --- (7 headers 0 lines) --- [2011-08-11 23:00:47] VERBOSE[17375] chan_sip.c: Transmitting (no NAT) to 65.55.174.100:5061: ACK sip:9...@um.outlook.com SIP/2.0 Via: SIP/2.0/TLS 1.2.3.4:5061;branch=z9hG4bK70149e02 Max-Forwards: 70 From: Test User sip:210@1.2.3.4;tag=as746bc17a To: sip:9...@um.outlook.com;tag=aprqngfrt-hm4td72c6 Contact: sip:210@1.2.3.4:5061;transport=TLS Call-ID: 535c9a2d211f19be5528b2ce7dbce0d4@1.2.3.4:5061 CSeq: 102 ACK User-Agent: FPBX-2.8.1(1.8.5.0) Content-Length: 0 --- [2011-08-11 23:00:48] VERBOSE[17000] chan_sip.c: Really destroying SIP dialog '535c9a2d211f19be5528b2ce7dbce0d4@1.2.3.4:5061' Method: INVITE TIA -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update
Re: [asterisk-users] Asterisk - Office 365 Unified Messaging... anyone done it?
does that mean you try setting dtmfmode=inband and made sure that 101 was no longer present in SDP? Still you got 488? good luck with that ;-) On Aug 16, 2011, at 1:04 PM, o o wrote: Alex, Thanks for the pointers. Digging through some Cisco documentation linked to as a guide for configuring CCM 8.0 with Office 365, it states that they support 711ulaw . I also tried setting dtmfmode=auto/rfc2833/info/inband with no luck. Trying to get someone with a brain at MS to work with me on this. From: Alex Vishnev alex9...@gmail.com To: o o bj_5...@yahoo.com; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, August 16, 2011 4:57 AM Subject: Re: [asterisk-users] Asterisk - Office 365 Unified Messaging... anyone done it? this could be an unsupported codec. Do you know if Office365 supports PCMU? I would also try to get rid of 101 (rfc2833) and see if that makes a difference On Aug 15, 2011, at 8:40 PM, o o wrote: Trying to make this work, and Office 365 support is useless, giving me the following response when I asked them for help troubleshooting a 488 Not Acceptable Here. Regarding your service request about configuring your PBX system with Office 365, we do not support specific setups for PBX systems for Unified Messaging. Please contact the vendor for more specific instructions and configurations. Here is a SIP debug: [2011-08-11 23:00:26] VERBOSE[17000] chan_sip.c: Reliably Transmitting (no NAT) to 65.55.174.100:5061: OPTIONS sip:um.outlook.com SIP/2.0 Via: SIP/2.0/TLS 1.2.3.4:5061;branch=z9hG4bK16884162 Max-Forwards: 70 From: Unknown sip:Unknown@1.2.3.4;tag=as438c582c To: sip:um.outlook.com Contact: sip:Unknown@1.2.3.4:5061;transport=TLS Call-ID: 67f260947dae7c27121ca30e5ee9d3ef@1.2.3.4:5061 CSeq: 102 OPTIONS User-Agent: FPBX-2.8.1(1.8.5.0) Date: Fri, 12 Aug 2011 06:00:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2011-08-11 23:00:27] VERBOSE[17375] chan_sip.c: --- SIP read from TLS:65.55.174.100:5061 --- SIP/2.0 200 OK Via: SIP/2.0/TLS 1.2.3.4:5061;branch=z9hG4bK16884162 From: Unknown sip:Unknown@1.2.3.4;tag=as438c582c To: sip:um.outlook.com;tag=b4ec76231 Call-ID: 67f260947dae7c27121ca30e5ee9d3ef@1.2.3.4:5061 CSeq: 102 OPTIONS ACCEPT: application/sdp CONTENT-LENGTH: 0 ALLOW: INVITE ALLOW: BYE ALLOW: CANCEL ALLOW: OPTIONS ALLOW: ACK ALLOW: INFO ALLOW: NOTIFY SERVER: RTCC/3.5.0.0 - [2011-08-11 23:00:27] VERBOSE[17375] chan_sip.c: --- (16 headers 0 lines) --- [2011-08-11 23:00:27] VERBOSE[17000] chan_sip.c: Really destroying SIP dialog '67f260947dae7c27121ca30e5ee9d3ef@1.2.3.4:5061' Method: OPTIONS [2011-08-11 23:00:47] VERBOSE[17999] chan_sip.c: Audio is at 5061 [2011-08-11 23:00:47] VERBOSE[17999] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [2011-08-11 23:00:47] VERBOSE[17999] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [2011-08-11 23:00:47] VERBOSE[17999] chan_sip.c: Reliably Transmitting (no NAT) to 65.55.174.100:5061: INVITE sip:9...@um.outlook.com SIP/2.0 Via: SIP/2.0/TLS 1.2.3.4:5061;branch=z9hG4bK70149e02 Max-Forwards: 70 From: Test User sip:210@1.2.3.4;tag=as746bc17a To: sip:9...@um.outlook.com Contact: sip:210@1.2.3.4:5061;transport=TLS Call-ID: 535c9a2d211f19be5528b2ce7dbce0d4@1.2.3.4:5061 CSeq: 102 INVITE User-Agent: FPBX-2.8.1(1.8.5.0) Date: Fri, 12 Aug 2011 06:00:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 238 v=0 o=root 1381221379 1381221379 IN IP4 1.2.3.4 s=Asterisk PBX 1.8.5.0 c=IN IP4 1.2.3.4 t=0 0 m=audio 17688 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [2011-08-11 23:00:47] VERBOSE[17375] chan_sip.c: --- SIP read from TLS:65.55.174.100:5061 --- SIP/2.0 100 Trying Via: SIP/2.0/TLS 1.2.3.4:5061;branch=z9hG4bK70149e02 From: Test User sip:210@1.2.3.4;tag=as746bc17a To: sip:9...@um.outlook.com Call-ID: 535c9a2d211f19be5528b2ce7dbce0d4@1.2.3.4:5061 CSeq: 102 INVITE Content-Length: 0 - [2011-08-11 23:00:47] VERBOSE[17375] chan_sip.c: --- (7 headers 0 lines) --- [2011-08-11 23:00:47] VERBOSE[17375] chan_sip.c: --- SIP read from TLS:65.55.174.100:5061 --- SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/TLS 1.2.3.4:5061;branch=z9hG4bK70149e02 From: Test User sip:210@1.2.3.4;tag=as746bc17a To: sip:9...@um.outlook.com;tag=aprqngfrt-hm4td72c6 Call-ID: 535c9a2d211f19be5528b2ce7dbce0d4@1.2.3.4:5061 CSeq: 102 INVITE Content-Length: 0 - [2011-08-11 23:00:47] VERBOSE[17375] chan_sip.c: --- (7 headers 0 lines) --- [2011-08-11 23:00:47] VERBOSE[17375] chan_sip.c: Transmitting (no NAT) to 65.55.174.100:5061: ACK sip:9...@um.outlook.com SIP/2.0 Via: SIP/2.0/TLS
Re: [asterisk-users] Queue agent login notification
you can monitor queue_log file for ADDMEMBER or REMOVEMEMBER events. from that point on, you can store them or take any other action. the other way is to use AMI an monitor for Agent login/logoff events On Aug 12, 2011, at 7:06 AM, Michael wrote: Hello, Is there a way to either store login/logout agent information in a database or at least send an email when an agent logs in or out of a queue? Thanks, Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No Audio after attended tranfer
No, that looks like a separate issue. Mine is a 100% repeatable and the asterisk does not lock up. SIP and RTP on other sessions are still going. in my cases this is the exchange I see Asterisk Service Provider INVITE (initial Invite to Service Provider with Outbound number) --- --200 OK -INVITE (put session on hold) --200OK --ACK -RTP -INVITE (no SDP) -- First transfer complete --200OK (SDP) --ACK -RTP -INVITE (no SDP) -- Second Transfer --200OK (SDP) --ACK (SDP) --RTP On Jul 19, 2011, at 3:41 AM, Stefan Schmidt wrote: Am 18.07.11 16:15, schrieb Alex Vishnev: I am wondering if anyone hit this case yet. I am using 1.6.2.19 and doing an attended transfer. The transfer is going to an outbound number (normally AA on another IP PBX). the audio on the first transfer is fine. But if the user requests a transfer from AA to another department, I loose audio from Asterisk to the 2nd transfer. Based on the review of SIP packets, the second transfer issues ACK+SDP. Anyone have experience with that? it looks like ACK+SDP is not being handled properly by asterisk. I searched thru JIRA cases, but did not find anything like that. Any help would be appreciated. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hello, maybe this is the problem you have: https://issues.asterisk.org/jira/browse/ASTERISK-18136 best regards Stefan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No Audio after attended tranfer
I am wondering if anyone hit this case yet. I am using 1.6.2.19 and doing an attended transfer. The transfer is going to an outbound number (normally AA on another IP PBX). the audio on the first transfer is fine. But if the user requests a transfer from AA to another department, I loose audio from Asterisk to the 2nd transfer. Based on the review of SIP packets, the second transfer issues ACK+SDP. Anyone have experience with that? it looks like ACK+SDP is not being handled properly by asterisk. I searched thru JIRA cases, but did not find anything like that. Any help would be appreciated. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RINGNOANSWER events in queue log
Does anyone know why i would get this RINGNOANSWER events in queue_log when clearly the agent is busy and call-waiting is disabled. 1309550595|1309550570.399965|2253|Local/05@from-internal/n|CONNECT|2|1309550593.399966|0 1309550632|1309550533.399961|2253|Local/11@from-internal/n|COMPLETECALLER|1|74|1 1309550663|1309550640.399969|2253|NONE|ENTERQUEUE||zz 1309550666|1309550640.399969|2253|Local/01@from-internal/n|CONNECT|3|1309550663.399971|0 //here it looks like Agent01 got the call. 1309550671|1309550648.399970|2525|NONE|ENTERQUEUE||zzz 1309550671|1309550648.399970|2525|Local/05@from-internal/n|RINGNOANSWER|0 1309550671|1309550648.399970|2525|Local/01@from-internal/n|RINGNOANSWER|0 // why is the system trying that channel for agent01 again? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue transfer order
Hello I have a small call center with about 7 queues. all agents are dynamic and they login to each queue via a dialplan. When you perform queue show you will see that all agents are able to service all queues. All queues have the same weight/priority. While monitoring a system I can see that callers with longer hold time can hang in the queue longer then new callers coming in. Agents are using rrmemory transfer method. The callers are in queue listening to music on hold and ready to be transferred. I am not using any announcements. How do I determine or enforce that callers with longer hold will go to an agent first? Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RINGNOANSWER IN queue_log
Does anyone know why i would get this RINGNOANSWER events in queue_log when clearly the agent is busy and call-waiting is disabled. 1309550595|1309550570.399965|2253|Local/05@from-internal/n|CONNECT|2|1309550593.399966|0 1309550632|1309550533.399961|2253|Local/11@from-internal/n|COMPLETECALLER|1|74|1 1309550663|1309550640.399969|2253|NONE|ENTERQUEUE||zz 1309550666|1309550640.399969|2253|Local/01@from-internal/n|CONNECT|3|1309550663.399971|0 //here it looks like Agent01 got the call. 1309550671|1309550648.399970|2525|NONE|ENTERQUEUE||zzz 1309550671|1309550648.399970|2525|Local/05@from-internal/n|RINGNOANSWER|0 1309550671|1309550648.399970|2525|Local/01@from-internal/n|RINGNOANSWER|0 // why is the system trying that channel for agent01 again? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bridged Call
I have a Bridged call with 2 parties. I want to redirect one party to a conference room and the other party to an outside number. I tried doing that with a dialplan. I used ChannelRedirect in the dialplan and redirected the first channel to the conference room. however, the second channel disconnects. Reading thru the mailing list i understand that this expected. However, I don't understand how I can connect the second channel to an outside number. Can someone give a hand? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ChannelRedirect
Hello, I am implementing a small ACD system on Asterisk 1.6.2.17.2 I need help with ChannelRedirect. I have a caller connected to an agent. The agent may request additional help by consulting another department. I can't use manual process with blind or directed transfer as the agent have many different numbers to dial. The message with the proper dial number is coming from the host. I got that handled in my application as well. but while I know the channels for agent and caller, I can't seem to get ChannelRedirect to work properly for me. I am using Dual ChannelRedirect with AMI interface by taking the caller port and directing the call to a predefined conference bridge. The other channel needs to be redirected to an outside number. For some reason, I have both channels going to the same number. I am not sure if I am specifying the right channels in ChannelRedirect. I am not married to AMI approach either. I can use AMI to Redirect channels to a dialing plan and handle everything in the dialing plan as well. It just seemed it was easy to use the dual ChannelRedirect. Please let me know what is the best way to handle this condition. I will also need to have an ability to conference the caller, agent and outside party if the agent requests that. It would be a great help to get the steps for that as well. thanks in advance. If I miss any crucial information, please let me know and I will post that Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Subscriptions
Doug, If you stop complaining and listen to what people are saying, you would be able to accomplish your goals. Some of your points have merit, but you are asking for help in all the wrong ways. Please remember, this list is for users, not developers. The user community is quite extensive in their backgrounds. Not all of us are developers or linux experts. There are people on the list that have used Asterisk and installed them in many enterprise environments, even though you claim it is not enterprise ready. My point is, instead of annoying everyone and triggering angry replies, you should change your tactic. Right now, you are not getting any useful information and flooding everyone's mail boxes with useless stuff. For example, you already heard that subscriptions in asterisk are work-in-progress. The entire project is work-in-progress. There are stable features and there are new features that are being worked on. If you respect the community, the community will respect you and give you what you need. I have been involved with asterisk for more then 1 year, and have nothing but good to say about people on this list and developers of asterisk.org in general. But, never... never... piss off these people, or you may as well quit your job and go do something else. There are a lot of experience on this list in Asterisk and Data Processing in general (i.e. people smarter then you are ;-). I think one of your mistake is that you trying to depend 100% on Asterisk to do the job. Asterisk is just a growing baby. It is growing fast, but still needs time. As a growing child, it has it challenges/problems and solutions. The solutions may not be very elegant and could only be temporary, but they are solutions to solve immediate business needs. Instead of complaining how inadequate redundancy is with asterisk, you should ask how to architect redundancy with asterisk. I have seen a number of solutions on the list regarding this. There were some that were done purely in asterisk and some were done using SER and Asterisk. Just to prove my point that there are people with solutions out in the community that are willing to help and share their experience, if you ask politely and with respect. Otherwise, you just get angry replies and people calling you nasty names. If you enjoy this, you can continue, but a lot of people will put you in their ignore lists and soon you will be talking to yourself. Ok... that's just my advice to you... you can take or leave it. But I strongly suggest you take it. It will make your job SO MUCH EASIER!!! Again, just my .2c Alex Message: 10 Date: Tue, 20 Dec 2005 20:52:55 -0700 From: Douglas Garstang [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] SIP Subscriptions To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=utf-8 Seems someone has some anger management issues. As I just stated in a previous post, it seems you have issues with me asking valid questions. I'm not sure why that is. The long email you rattled off with all my questions where quite valid. Your issue with that is.? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Satellite WAN
Adam, I personally think that replacing hard-wired network and going with Sats is a mistake. Judging from pure round-trip delay you measured the packet round trip seems sufficient to have a good conversation, but pinging is not enough to trouble shoot the network problems. You will need to do a lot more work to identify the problem with this location. If both locations are under your control, then I would put network probes in both places to identify exactly when and how the quality problems appear. Network probes would identify the type and the amount of traffic both sides are sending and receiving. There are network probes that can even do Voice Quality Analysis and determine how well your network is performing. As a side step, I would also look at internal location in New Brunswick, because that is the only location you are having problems with. I would check to see if there are simple network problems like bad network port, network card, packet collision on the network, network card on routers, etc. I am sure you have already considered simple things like that, however you need to methodically go thru each one to see where the problems are. Replacing the network would be my last alternative. If you are at that point, well then just ignore this email. Otherwise, there are plenty of things you can do before taking such a drastic measure. HTH Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Robins Sent: Wednesday, November 02, 2005 12:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Satellite WAN We have built an Asterisk network using an MPLS-based IP VPN. We have one location in New Brunswick Canada that consistently gives us major quality problems, whereas the others are flawless. Quality problems take the form of static, poor voice tonality, popping clicking, drops, sporadic echo, you name it. The latency of a QoS prioritized packet between the Canada site and our hub in Atlanta is 85ms (ping). I have been searching for an alternative network provider, but I'm told that they would all take the same route from the US into Canada, as there is simply no major backbone running into NB east of Toronto. So now I'm thinking about satellite. I have no idea if a) this would even be economically feasible, and b) if the latency would be any better. If anyone out there has had any such satellite network experience with VoIP, I like to hear from you. Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] context question
I briefly looked thru the code and I don't believe there is a way to separate the context or really make them independent. I know exactly what you want to accomplish. I think it could be done with a little trick. For example, every customer on hosted pbx would be given some kind of unique identifier. The back-end would silently place the identifier at the beginning or the end of the context making the new name totally unique. The front-end would hide identifier from users view and just present the name of the context. That way, customers can name their context anything they like and there would be no collision. In that case, Goto would also be local to the context as the real context name will contain customer id. Does that work for you? Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of trixter http://www.0xdecafbad.com Sent: Friday, September 23, 2005 11:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] context question They are aware of each other in 2 senses. First you can goto() them. I wanted to stop the ability of someone to put in a goto() in their dialplan to a context that is someone elses (think asterisk hosting). Second naming collissions. I wanted to stop two people from having the same name and causing grief that way. That is why I made the references about prepending some customer id or something, but I dont think that is the best way to accomplish this (personal preference), so it will either be an AGI to accomplish this or it will be something else that already exists that I havent been able to locate as yet. On Fri, 2005-09-23 at 21:50 -0500, [EMAIL PROTECTED] wrote: I may be missing something, but aren't all contexts unaware of each other be default? If I do the following [contexta] exten = 3200,1,Dial(SIP/3200,5) [contextb] exten = 3300,1,Dial(SIP/3300,5) Each context has a phone and they can't call each other. The are completely isolated. Unless I'm missing what you are trying to do trixter http://www.0xdecafbad.com wrote: Is there any way within asterisk to limit the scope of contexts, basically to make one context totally unaware of another. The application I had in mind involved allowing users to create their own dial plans. To that end I wanted to make it so that a given user could not call a different users dialplan. I could filter everything and prepend a customer id to every context they specify, but that can get ugly fast, especially when the parser misses something. If this doesnt exist I can surely do it with an agi, and that is the road I am headed down right now, but why duplicate an effect that may already exist? Thanks. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] custom ring tone
Actually that is not true. You can have a short time where audio path is open prior to answering of the call. This depends on the provider, switch and software. I think the largest window I have seen is 90 seconds. Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of trixter http://www.0xdecafbad.com Sent: Thursday, September 22, 2005 4:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] custom ring tone On Thu, 2005-09-22 at 22:01 +0200, Marko Rakar wrote: yes, yes the thing is that local telco uses this feature for their customer support line and also one of wireless providers now also offers ability to customize your ring tone I was told that if you have analog or even ISDN BRI line that ring tone is generated in your local teclo exchange, but if you have connection like E1 that it is generated localy in your PBX (explanation being that So in short you can have a toll free info line without actually paying for the toll free. While its not interactive, by not sending answering supervision the caller is not charged. Interesting concept they have there, sure beats the 10k resistor trick from the analog switch days (although then you could talk to the other person). -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] custom ring tone
Yes, sometime audio is both ways. Sometimes, it is just one way. This only works in digital network (T1/E1 to T1/E1 (CAS handoff) - the window is much shorter even in feature group D. or T1/E1 to T1/E1 PRI signaling where the window could be as large as 90 seconds. Again, that depends on country, provider, switch software. You can't get this if you are calling POTS lines. Hth Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of trixter http://www.0xdecafbad.com Sent: Thursday, September 22, 2005 9:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] custom ring tone Audio both ways? Sure would beat the collect call game :P On Thu, 2005-09-22 at 21:15 -0400, Alex Vishnev wrote: Actually that is not true. You can have a short time where audio path is open prior to answering of the call. This depends on the provider, switch and software. I think the largest window I have seen is 90 seconds. Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of trixter http://www.0xdecafbad.com Sent: Thursday, September 22, 2005 4:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] custom ring tone On Thu, 2005-09-22 at 22:01 +0200, Marko Rakar wrote: yes, yes the thing is that local telco uses this feature for their customer support line and also one of wireless providers now also offers ability to customize your ring tone I was told that if you have analog or even ISDN BRI line that ring tone is generated in your local teclo exchange, but if you have connection like E1 that it is generated localy in your PBX (explanation being that So in short you can have a toll free info line without actually paying for the toll free. While its not interactive, by not sending answering supervision the caller is not charged. Interesting concept they have there, sure beats the 10k resistor trick from the analog switch days (although then you could talk to the other person). -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Benchmarking / Stress Testing
sipsak (www.sipsak.org. ) is an excellent tool for this. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sherwood McGowan Sent: Friday, August 26, 2005 10:48 AM To: 'Asterisk Developers Mailing List' Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] SIP Benchmarking / Stress Testing Anyone have a good tool(s) to use for simulating a bunch of calls? Benchmarking or stress testing? I only need SIP protocol, and do appreciate any replies...I realize I could google it, but I am looking for opinions as well. Sherwood McGowan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can't bridge between h323 and sip calls
Hello, I am using asterisk CVS-head from 6/28. I am also using chan_oh323 that comes with asterisk. I tried to place a call from h323 device into asterisk. in extensions.conf, I routed the call to my sip phone. The sip phone was already registered with asterisk. all the signaling looks ok to me. The sip phone rings when h323 call hits the asterisk box. But then the call is dropped. It appears that asterisk is trying to convert incoming g.729 codec to ulaw and it can't. I was assumed that g.729 will pass-thru to the phone. In fact, when an invite is sent bothg G729, G723 are codecs in SDP. However, when SIP phone answers, it only replies with g723 on 200OK. I am still unclear about that, but that's not really that important. I would like to find out why I can't bridge these two legs. below is the trace from the call. I am suspecting that a line below is the cause, but not sure why. Can someone help??? Jun 29 10:59:46 WARNING[8862]: app_dial.c:1324 dial_exec_full: Had to drop call because I couldn't make H323/ip$64.243.115.153:32971/11679 compatible with SIP/debit-9f37 -asterisk log-- -- Executing Dial(H323/ip$64.243.115.153:32971/11679, SIP/debit|20|rt) in new stack Jun 29 10:59:41 NOTICE[8862]: channel.c:1893 set_format: Unable to find a path from g729 to ulaw Jun 29 10:59:41 NOTICE[8862]: channel.c:1893 set_format: Unable to find a path from g729 to ulaw We're at 64.243.115.157 port 18192 Answering with capability 0x1 (g723) Answering with capability 0x4 (ulaw) Answering with capability 0x8 (alaw) Answering with capability 0x100 (g729) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 13 lines Reliably Transmitting (NAT) to 69.115.205.168:4152: INVITE sip:[EMAIL PROTECTED]:4146 SIP/2.0 Via: SIP/2.0/UDP 64.243.115.157:5060;branch=z9hG4bK5aab56d3;rport From: 7323600296 sip:[EMAIL PROTECTED];tag=as492d969f To: sip:[EMAIL PROTECTED]:4146 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 29 Jun 2005 14:59:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 292 v=0 o=root 8862 8862 IN IP4 64.243.115.157 s=session c=IN IP4 64.243.115.157 t=0 0 m=audio 18192 RTP/AVP 4 0 8 18 101 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called debit Jun 29 10:59:41 WARNING[8862]: chan_h323.c:588 oh323_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4) Jun 29 10:59:41 NOTICE[8862]: channel.c:1893 set_format: Unable to find a path from g729 to slin Jun 29 10:59:41 WARNING[8862]: indications.c:99 playtones_alloc: Unable to set 'H323/ip$64.243.115.153:32971/11679' to signed linear format (write) voip*CLI -- SIP read from 69.115.205.168:4152: SIP/2.0 100 Trying Via: SIP/2.0/UDP 64.243.115.157:5060;branch=z9hG4bK5aab56d3;rport From: 7323600296 sip:[EMAIL PROTECTED];tag=as492d969f To: sip:[EMAIL PROTECTED]:4146 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Grandstream BT100 1.0.5.16 Warning: 399 69.115.205.168 detected NAT type is symmetric NAT Content-Length: 0 --- (9 headers 0 lines)--- voip*CLI -- SIP read from 69.115.205.168:4152: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 64.243.115.157:5060;branch=z9hG4bK5aab56d3;rport From: 7323600296 sip:[EMAIL PROTECTED];tag=as492d969f To: sip:[EMAIL PROTECTED]:4146;tag=2cfc88182690d7d1 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Grandstream BT100 1.0.5.16 Warning: 399 69.115.205.168 detected NAT type is symmetric NAT Content-Length: 0 --- (9 headers 0 lines)--- -- SIP/debit-9f37 is ringing voip*CLI -- SIP read from 69.115.205.168:4152: SIP/2.0 200 OK Via: SIP/2.0/UDP 64.243.115.157:5060;branch=z9hG4bK5aab56d3;rport From: 7323600296 sip:[EMAIL PROTECTED];tag=as492d969f To: sip:[EMAIL PROTECTED]:4146;tag=2cfc88182690d7d1 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Grandstream BT100 1.0.5.16 Warning: 399 69.115.205.168 detected NAT type is symmetric NAT Contact: sip:[EMAIL PROTECTED]:4146 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Supported: replaces Content-Length: 213 v=0 o=debit 0 8000 IN IP4 69.115.205.168 s=SIP Call c=IN IP4 69.115.205.168 t=0 0 m=audio 4192 RTP/AVP 4 101 a=sendrecv a=rtpmap:4 G723/8000 a=ptime:30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 --- (13 headers 11 lines)--- Found RTP audio format 4 Found RTP audio format 101 Peer audio RTP is at port 69.115.205.168:4192 Found description format G723 Found description format telephone-event Capabilities: us - 0x10d (g723|ulaw|alaw|g729), peer - audio=0x1 (g723)/video=0x0 (nothing), combined - 0x1 (g723) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Jun 29 10:59:46 NOTICE[8862]: channel.c:1893 set_format: Unable to find a path from g723 to ulaw Jun
RE: [Asterisk-Users] Argentina and Mexico DID's Termination
Charlie, I am interested. Can you contact me off-list with details. Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos A Maimone @ GAUSS Sent: Thursday, June 23, 2005 3:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Argentina and Mexico DID's Termination Anyone interested in Mexico and Argentina DID's and termination? It's for exchange Thanks, Charlie Maimone ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CTI
You may also want to check the following link http://www.voip-info.org/wiki-MSN%20PHP. This is work in progress, but I think it may help you. it is based on IM messaging protocol to/from MSN Messenger. I don't believe there is a redirect to hard phones, but I think that could be part of command dictionary. HTH Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Tuesday, May 24, 2005 10:54 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] CTI Nevertheless Mozphone looks like a great softphone and the manager windows, etc gave me some great ideas! |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Jean-Denis Girard |Sent: Martes, 24 de Mayo de 2005 06:20 p.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] CTI | |Anton Krall a écrit : | If I have a hardphone, can mozphone redirect the call to my |hardphone | instead of using the softphone? For example, dial using the pc, see | callerid on the pc, etc but if I answer the call, redirect to my | hardphone? Or when making calls, send them to my hardphone? | |Sorry I didn't understand you wanted to use a hardphone. |MozPhone is obviously a softphone only solution. | | |Thanks, |Jean-Denis |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] two isdn cards
Mike, The cable needs to be a cross-over cable when connecting directly between 2 T1s, bypassing PSTN. One side of isdn has to be configured as TE and the other as NT. Only 4 wires are needed (not full 8 wires) to build a T1 cross-over. If you are connecting the systems thru pstn, you need regular T1 cable. Also, please remember to configure timing/clocking on both systems. If you are connected to pstn, then you will need to configure slave clocking on your side. If you are connecting 2 systems without pstn, then one must generate clock and another slave clock. Without proper t1 clocking you will see frame slips and errors on t1 line. ISDN is very sensitive to clocking, while regular RBS t1 can function with with frame slips, except you may hear pops/clicks or missed in your audio stream. HTH Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stankiewicz Michael Sent: Monday, May 23, 2005 11:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] two isdn cards thanks a lot, i've googled around hunting for an answer to my biggest doubt: the cross-cable. i understand that it looks like an cat-5 cross-cable and how it has to be done, but ... why 8 wires ? i found this image: http://www.gcom.com/home/support/t1crossover.html and that one: http://www.voip-info.org/wiki-crossover+T1+cable?page=crossover%20T1%20cable comments_threshold=0comments_offset=0comments_sort_mode=commentDate_desc ... so: the cable that goes from the isdn NT to asterisk should be an 8 wires isdn-cross-cable ? thanks for those newbye delightenments :) ciao mike On Mon, 2005-05-23 at 16:50, Emanuele Pucciarelli wrote: Stankiewicz Michael wrote: i followed this how-to: http://www.voip-info.org/tiki-index.php?page=Asterisk%20zaphfc%20install26 having in response no sign of life. If the module doesn't even get installed, or the kernel does not report any card as recognized, you could tweak the initialization routines to add PCI IDs for your own cards, and hope they work correctly. If the cards are recognized, there should be nothing to worry about: either they work with zaphfc or they don't, modulo interrupt troubles. the software side is pretty straightforward but i have many doubts on the hardware deployment: 1- the idsn cable going from asterisk to the NT sould be a cross cable ? Yes. But not an Ethernet cross-cable, an ISDN cross-cable; there's a pointer on the wiki to a page on isdn.jolly.de explaning how the cable should be made, and suggestions about how to take advantage of a disused NT. I reckon that telephony folks call it an ISDN TX/RX cable. 2- it should have 100 ohm resistors (if yes, where can i find the schemes )? Yes, the bus should be terminated (so the resistors don't have to be on the cable itself). A full description of the bus is in the ETSI standard for ISDN layer 1 (www.etsi.org). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP/nat situation
Pizco, SER is definitely better suited to deal with NAT issues then ASTERISK is. I suggest looking at SER and NAT helpers like media proxy application (part of SER). I also recommend looking at NAT devices at SER wiki page to make sure that your router/nat device is compatible. In general, this is doable, but will require a lot of playing around to get it right. There are a lot of threads on both SER and ASTERISK wiki site to get both working nicely together. Asterisk/SER Wiki Site www.voip-info.org HTH Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pizco Dominguez Sent: Wednesday, May 18, 2005 8:23 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SIP/nat situation Hi. We are trying to set up asterisk to service a wireless community in our town. We have about 30/40 wireless working nodes each one with a 10.34.x.x/24 subnet for users. Each one of these addresses can potentially have a 192.168.x.x/x subnet. On top, the wireless nodes, themselves, are linked in 172.16.x.x/x subnets. On top of the top, there is internet and cool things for people, like iptel, fwd, etc. If there is SIP paradise, our set up is most definitely nearer to hell, regarding nat, because no one knows which kind of address the asterisk client is going to come up with. The more I fiddle with asterisk and read this list, the bigger my doubts about the possibility of making asterisk (SIP) work for most of us (it already works for some). A friend suggested that maybe putting up one or two asterisk boxes to work and using SER in strategically choosen nodes we could get away with it. I'm having a look at SER and think that maybe it could work for us, but wanted to check with some other people before diving into the unknown. Answers like Give it a try, Don't even think of it or Better back to tam-tam and smoke signalling are wellcome. Thanks for your time. -- Pizco Dominguez -- -- GPGKEY: gpg --keyserver pgp.rediris.es --recv-key 8DE37A4D FINGERPRINT:85CB 4323 F322 5837 EDB5 2033 6FB2 C326 8DE3 7A4D -- -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Which free calling card app most suitedforcommercial use?
I think the word crap is a pretty strong word and is not fare to the authors. Everyone have their own requirements of how billing should or should not work. Everyone is exposed to a different way a pre-paid calling card platform should behave. I have been in pre-paid environment for almost 15 years and seen/implemented some interesting business models. All of them depend on the provider and how the product is brought to market. Point is, there will never be a unified billing system that will satisfy every requirement of every pre-paid carrier in the world. I think these guys did a good job showing the community how pre-paid billing should be implemented and interfaced with asterisk and therefore deserve a credit for that. If you don't like the way they implemented things, then contribute extensions or patches. If you don't like the architecture and don't think a particular approach can be extended, then contribute your own work and show everyone that it is better. Until you do, avoid the words like crap when referring to other people efforts. Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kanuri, Seshu (Company IT) Sent: Wednesday, April 20, 2005 9:46 AM To: snacktime; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Which free calling card app most suitedforcommercial use? My opinion is that both are Crap. Both of them have a flaw in their base design, which is difficult to explain in a post like this. Suffice to say that these two applications neither support nor designed for mutilpe routes ( multiple Area codes with Destination groups) nor multiple rate plans(Provider rates or buying rates and selling rates) nor multiple business models(retail, wholesale, corporate customers) Hence both of them cannot be the base for a commercial grade billing system for a Calling card Model. These apps canot be used for a realtime call control using CPD (Call Progress Detection) and Prepaid amounts for a post-paid Billing and call disconnect. Without this very essential feature for a commercial Calling card billing application, you would be better off calculating the calls from the Master.csv file for a post paid bill management. AreskiCC is a little more thought-driven and hence can be improved upon. If anyone is interested in developing a full fledged billing system, I have created a deisgn document ( a very elaborate rough draft infact) which I can share with you. Seshu Kanuri -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of snacktime Sent: Tuesday, April 19, 2005 5:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Which free calling card app most suited forcommercial use? I'm working on an * billing system, and instead of reinventing the wheel I would prefer to use an existing codebase for the calling card portion. The two that look most promising are astcc and the * prepaid billing application that uses postgresql. Any comments? Chris NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Distributed organizations - large scale public sector rollout
Eivind Most obvious solution is snmp. Using snmp you can collect statistics and provision your remote systems. However, SNMP is an enabler and not the full solution. You still need to write SMUX agents and develop application MIBS that allow you to get/store application specific data. To my knowledge Asterisk does not support any MIB reporting to date. You will need to extend asterisk with scripts and applications to provide you the data. Most of scripting tools like perl or php have good support for SNMP. HTH Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eivind Trondsen Sent: Monday, April 18, 2005 5:26 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Distributed organizations - large scale public sector rollout Hi List I am working with a pilot project for a Norwegian regional government to evaluate Asterisk for a large number of sites and users. The goal of the project is to have a unified VoIP-system to replace the disorganized collection of legacy PBX in use today. By distributed organization I mean an organization that consists of many, dispersed units, each with it's own existing telephony system, and with distinct number series. The goals of a unified system are several: - Lower traffic cost through a common backbone between sites and a common exit-point to the PSTN (either via IP or legacy lines). - Lower admin cost through unified, centralized management. - Added value through rollout of applications (voicemail, conferencing, IVR) that become globally available in the system. My main concern is manageability. From what I have seen of the available management tools there are none that address the needs of a distributed system. They all seems aimed at the SMB market, and don't leverage resources such as LDAP directories. Does anyone have any experience with management tools for Asterisk in a similar scenario? I am also very interrested in getting in touch with people working in similar projects. There is a large political element in rolling out Open Source telephony on such a scale, and having a network of similar projects could be of great value. I hope to be able to post to this list on our progress. Best regards -- Eivind Trondsen Wingnut Information Systems Norway ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] BT100 wrong NAT detection
Tomas, Yes, BT100 is a little picky on the use of Stun Servers. For example, it will not work at all with Vovida Stun server. Also, Stun negotiation takes some time. So if you rebooted the phone, I would suggest waiting 15-30 seconds until phone syncs up with Stun server and requests binding. You can also run ethereal on your LAN and monitor the packets coming from Bt100. Then you can compare them to Xlite or other phones to see how they differ. I would also suggest contacting grandstream and getting the latest firmware for granstream. Another thing that made we wonder is when you said you are running Stun on the same system as asterisk. Normally Stun requires 2 systems, system with 2 NIC cards, or at minimum 2 IP addresses that stun can bind to. Is that what you are doing? Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomas Florian Sent: Saturday, April 16, 2005 1:17 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] BT100 wrong NAT detection Hello, I'm having trouble getting BT100 to identify NAT type reliably for Asterisk. My setup is as follows: - Asterisk is on the open internet 142.x.x.41 - BT100 phones are behind NATs - I use STUN for my BT100 : 142.x.x.41 (same server as Asterisk) - BT100 firmware (tried .16,.18,.23 same results) When I go to the BT100 setup page I can see the following: - detected NAT type is symmetric NAT OR (sometimes) - detected NAT type (blank) Both of these are wrong as my NAT type should be: Port restricted NAT ... if I'm lucky sometimes BT100 comes back with port restricted answer and in that case I'm ready to go .. but it rarely works after a reboot ... sometimes yes sometimes no .. I tested the STUN server and my actual NAT type by running the WinSTUN ... it always answers correctly 100% of the time. I also tried setting the BT100 STUN server to some public STUN servers .. no luck. ... so why is BT100 so unreliable??? I even did ./sever -v to watch my STUN server in action and it does actually talk to the BT100s on every phone reboot .. but the weird thing is that between BT100 and STUN there are only 3 messages sent whereas between XLite and STUN or WinStunClient and STUN server there are about 8+ ... it's almost as though BT100 gives up .. is BT100 compatible only with certain STUN servers? Is there some trick to this? What else can watch to troubleshoot this situation? Thank you, Tomas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] BT100 wrong NAT detection
Tomas, There is mystun on sourceforge, but I think the only way to down load it is to build it from cvs source. I normally use public stun servers from grandstream or xten. Hth Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomas Florian Sent: Saturday, April 16, 2005 2:23 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] BT100 wrong NAT detection One more question ... I did a search on Google for STUN servers and didn't find any other open source server other than Vovida's What other open source Stun servers are there? And if there are none, what commercial one have you found to work well with BT100? Thanks again, Tomas -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Vishnev Sent: Saturday, April 16, 2005 12:00 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] BT100 wrong NAT detection Tomas, Yes, BT100 is a little picky on the use of Stun Servers. For example, it will not work at all with Vovida Stun server. Also, Stun negotiation takes some time. So if you rebooted the phone, I would suggest waiting 15-30 seconds until phone syncs up with Stun server and requests binding. You can also run ethereal on your LAN and monitor the packets coming from Bt100. Then you can compare them to Xlite or other phones to see how they differ. I would also suggest contacting grandstream and getting the latest firmware for granstream. Another thing that made we wonder is when you said you are running Stun on the same system as asterisk. Normally Stun requires 2 systems, system with 2 NIC cards, or at minimum 2 IP addresses that stun can bind to. Is that what you are doing? Alex -Original Message- rom: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomas Florian Sent: Saturday, April 16, 2005 1:17 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] BT100 wrong NAT detection Hello, I'm having trouble getting BT100 to identify NAT type reliably for Asterisk. My setup is as follows: - Asterisk is on the open internet 142.x.x.41 - BT100 phones are behind NATs - I use STUN for my BT100 : 142.x.x.41 (same server as Asterisk) - BT100 firmware (tried .16,.18,.23 same results) When I go to the BT100 setup page I can see the following: - detected NAT type is symmetric NAT OR (sometimes) - detected NAT type (blank) Both of these are wrong as my NAT type should be: Port restricted NAT ... if I'm lucky sometimes BT100 comes back with port restricted answer and in that case I'm ready to go .. but it rarely works after a reboot ... sometimes yes sometimes no .. I tested the STUN server and my actual NAT type by running the WinSTUN ... it always answers correctly 100% of the time. I also tried setting the BT100 STUN server to some public STUN servers .. no luck. ... so why is BT100 so unreliable??? I even did ./sever -v to watch my STUN server in action and it does actually talk to the BT100s on every phone reboot .. but the weird thing is that between BT100 and STUN there are only 3 messages sent whereas between XLite and STUN or WinStunClient and STUN server there are about 8+ ... it's almost as though BT100 gives up .. is BT100 compatible only with certain STUN servers? Is there some trick to this? What else can watch to troubleshoot this situation? Thank you, Tomas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DTMF does not work with g729 and AGI
First of all I hope you realize you can't have the same context activated at the same time for the same host as * does not support this. So I am just thinking the configuration below are just examples of what you tried. I strongly suggest using dtmfmode=rfc2833 and dtmfmode=info instead of inband. Inband will only work for g711 as there is no compression. Secondly, I would suggest looking at your client and configure the client to match * config. If that does not work, I would capture the data with ethereal and decode the protocol to see what is happening. Most likely problem is with your client. Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, April 14, 2005 8:11 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] DTMF does not work with g729 and AGI Hello, I have an AGI script that runs a menu at two levels of a tree. If I call the extension from a voip phone with g711, the menu works fine and accepts DTMF no probs. Then, when I Call from a DID, it sends call using SIP and g729 to¨* box. The IVR also starts running, but no DTMF is deteced. I have tried various configs (combinations of dtmfmode=info, dtmfmode=rfc2833 and dtmfrelax=yes, dtmfrelax=no) with no success. Any hint? sip.conf [SS_SIP] type=peer host=XXX.XX.XXX.XX dtmfrelax=no ;dtmfmode=rfc2833 dtmfmode=info context=outbound disallow=all allow=g723.1 allow=g729 [SS_SIP] type=user host=XXX.XX.XXX.XX context=outbound dtmfmode=inband disallow=all allow=g723.1 allow=g729 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk became berserk when Internet connection is down and can't register to SIP server.
I think there are a couple of things you can do: 1. Switch the provider to get a stable internet connection ;-) 2. convert your lookups to IP addresses instead of domains. However, if you clients register with address like [EMAIL PROTECTED], then dns will be used to resolve blah.com and then you have a problem. I am not sure if converting to ip addresses is doable on a large scale. 3. monitor your internet connection with another script. If the connection fails then automatically edit * config file to remove your registration with FWD and reload the proper config. 4. configure * with realtime extensions and place peers into mysql db. Then use option 3 to monitor your internet connection and remove the peer on failure. This step does not require reloading config. HTH Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kong Sent: Thursday, April 14, 2005 11:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk became berserk when Internet connection is down and can't register to SIP server. So, any way i can resolve this problem? At 10:55 AM 4/15/2005, you wrote: On 4/14/05, Kong [EMAIL PROTECTED] wrote: Hi, i found a case here, i really don't know is it a bug or something else. i have like 200 ip phones connected to my * server, (ATA's and softphones). and i had it register to SIP service (FWD), so, when my internet connection is down, * is not able to register itself to FWD, never mind that, but it made all the extension berserk. all the client are not able to login to the server. error msg is login timeout, but once i remark the register = :[EMAIL PROTECTED] and restarted the server, immediately * became back to normal. so, i was wondering, is the a bug or something? coz my internet provider is not consistent, sometimes it goes down. thank you. I seem to remember a bug like this that had to do with dns lookups I think. Maybe someone else can remember the exact details and what version it was in. Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk behind NAT
Try setting externip=(asterisk public ip address) Hth Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Oswaldo Arratia Sent: Friday, April 15, 2005 12:56 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Asterisk behind NAT Hi List, I've spent hours researching on this topic, found tons of info, so far it doesn't work yet. Here's the scenario Asterisk box connected to a router (DMZ enabled to Asterisk) and trying to send calls to an outside provider. My SIP phones (outside * NAT) are able to register with no problem. The problem comes when I send a call out to my provider I get no audio in either way. My provider sees only my private IP. How can I send my provider the public IP?? I have in my sip.conf: [general] port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) nat=yes externip = x.x.x.x localnet=192.168.1.0/24 (remember I enabled DMZ on the router so all ports are being forwarded to *) Despite of what I do, the externip and the localnet my provider only gets my * private IP. Any tip how to solve this?? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk behind NAT
Can you show your outbound peer configuration? If you are registering, please include that as well. Thanks Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Oswaldo Arratia Sent: Friday, April 15, 2005 9:44 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk behind NAT I have... Externip=x.x.x.xand nothing... Does not seem to help in anything. Still my provider sees the private IP. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Vishnev Sent: Friday, April 15, 2005 9:05 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk behind NAT Try setting externip=(asterisk public ip address) Hth Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Oswaldo Arratia Sent: Friday, April 15, 2005 12:56 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Asterisk behind NAT Hi List, I've spent hours researching on this topic, found tons of info, so far it doesn't work yet. Here's the scenario Asterisk box connected to a router (DMZ enabled to Asterisk) and trying to send calls to an outside provider. My SIP phones (outside * NAT) are able to register with no problem. The problem comes when I send a call out to my provider I get no audio in either way. My provider sees only my private IP. How can I send my provider the public IP?? I have in my sip.conf: [general] port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) nat=yes externip = x.x.x.x localnet=192.168.1.0/24 (remember I enabled DMZ on the router so all ports are being forwarded to *) Despite of what I do, the externip and the localnet my provider only gets my * private IP. Any tip how to solve this?? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RTP not being sent by asterisk
Can you capture Ethernet traffic with ethereal or similar tools and show what is happening? Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of trixter http://www.0xdecafbad.com Sent: Thursday, April 14, 2005 1:56 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] RTP not being sent by asterisk I am having an odd problem that started somepoint in the last couple days with no known config change. Asterisk will receive RTP data but will not send it. If someone calls my asterisk box, it will hang on any Playback() or Background() call. No data is ever sent on the RTP stream, verified with a packet sniffer. I disabled all bandwidth shaping and firewall settings while testing which had no effect on resolving this. SIP traffic goes back and forth, and a sip debug shows everything being set up. I have deinstalled and reinstalled what was previously working. A friend who has the same version installed from the same place has no problems with his setup. I started with asterisk from debian testing however built from CVS a few minutes ago and have exactly the same problem. I am now stuck on where to look next to find the problem and need to get my asterisk system working again quickly. Any ideas would be greatly appreciated. Sample I called from extension.conf exten = 123,1,answer exten = 123,2,wait,2 exten = 123,3,playback(beep) ; it hangs on this beep exten = 123,4,playback(beep) exten = 123,5,playback(beep) exten = 123,6,hangup sip.conf was not changed at all, and that works for in/out. The only problem I have is people dialing into my asterisk box, the applciations run, DTMF is read, callers just get absolutly no prompts. -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 881 8487 FreeWorldDialup: 635378 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Strange intermittent NAT problem with BT100s
I have seen the same problem as well. If don't think this is a problem with BT100. I think the problem is with public STUN server. I think sometimes, the server is too overloaded and can't provide the translation. That is when you are getting the problem with your clients behind NAT. the only solution is to build your own stun server and use it, instead of using public servers. Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomas Florian Sent: Thursday, April 14, 2005 2:32 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Strange intermittent NAT problem with BT100s Hello, I have a strange problem whenever I have 2 or more BT100s behind NAT. I am not able to reproduce this error reliably, but it happens every 2-5 minutes. The general setup is that there is Asterisk server sitting at a central location. Some peers connect directly (206,205,201) but some (204,203,200) connect through NAT. This all works fine ...but it is extremely unreliable ... I get UNREACHABLE and then OK again ... UNREACHABLE and OK again .. unpredictably. When it's OK I can make phone calls no problem of course when it goes UNREACHABLE there is trouble. I tried to replace one of the BT100 phones with X-Lite and that one is OK (~40ms) rock solid - or seems to be so far. So it seems that there is something weird going on with BT100 My configuration of BT100 is as follows: - firmware 1.0.5.23 (I've noticed similar problems with .16 also) - detected NAT type is symmetric NAT - STUN stun.xten.net (I'm using Xtens ... or do I have to use my own???) - no outbound proxy - register expiration = 1 - keep alive interval = 20 sec (I also tried as low as 1 sec) My sip configuration uses: - nat = yes - qualify = yes (I also tried longer qualify 1 with no luck) This is what I get with sip show peers ... the 204 and 200 are BT100 and sometimes one or both go UNREACHABLE for a while ... 203 is X-Lite and didn't go UNREACHABLE yet. 206/206 (Unspecified)D 255.255.255.255 0 Unmonitored 205/205 (Unspecified)D 255.255.255.255 0 Unmonitored 204/204 209.x.x.125 D N 255.255.255.255 38340OK (40 ms) 203/203 209.x.x.125 D N 255.255.255.255 1548 OK (38 ms) 201/201 192.168.2.112D 255.255.255.255 5060 Unmonitored 200/200 209.x.x.125 D N 255.255.255.255 37838 UNREACHABLE Any ideas? Is there some trick to get BT100 to cooperate? Thanks, Tomas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] trying the xc-ast queue_log analyzer
The demo does not seem to be working, I am doing something wrong. It is constantly complaints that file placed in 'File' field is not found. Please let me know how to resolve this. Thanks Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of lenz Sent: Thursday, April 14, 2005 2:36 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] trying the xc-ast queue_log analyzer Hello list, I am glad to announce that it is now possible to try XC-AST, the queue_log file analyzer implementing most call centre metrics for the app_queue, using a demo password. See http://demo.xcept.it/xc-ast/xcast-live.jsp Some people complained that it was quite too complex to set up a servlet engine and a database just to check how XC-AST worked, so we thought that it could be nice if you could simply try and run it with no strings attached. If you have actual data you'd like to try XC-AST on, like something from your existing queues, we can set it up on a private area of the demo system so you can see how the system behaves. XC-AST is a commercial product, but is free for smaller installations, like SOHOs and home hackers. Please keep in mind that the demo server is a rather low-power one, so it coould be much less responsive than an actual production machine on your LAN. :-) Bye for now, l. -- Assum est, versa et manduca. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voice controlled calling?
Magnus, As far as I remember, Festival is only Text-to-speech, not voice recognition. In order to do what you want you need a voice recognition application. Also, compression gives voice recognition quite a challenge, as the speech samples arriving at the voip voice recognition engine is not the same as it was spoken using regular 64kbits pstn connection (as an example). Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of magnus Sent: Thursday, April 07, 2005 4:35 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Voice controlled calling? Hello all, rumours reach me of a way that the UK incumbent operator is planning to compete with VOIP by offering voice activated dialling, e.g. pick up the handset and through speech dial from your personnel directory, this leads me to wonder if this could be performed with Asterisk and Festival? I have looked in the WiKi and goggled, but can find no information on if this is possible, (particularly with SIP?) hence this question, has anyone achieved this? Intent would be to make is simple for non technical person - E.g. Grandma picks up the phone, does not have to worry about entering any digits and then makes call by voice control - for example call daughter etc. The key here is not to need any human interaction with the phone, other then picking up handset, the rest controlled by voice. Many thanks Magnus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP phones to Asterisk using MAC addressinsteadof IP address
If you setup host=dynamic in sip.conf, then the registration does not depend on ip address. It depends on sip user name of sip URI. You need to provision sip user name inside each phone. Please bare in mind that it is different then sip authentication name. Hope this helps Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giles Coochey Sent: Monday, April 04, 2005 10:33 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] SIP phones to Asterisk using MAC addressinsteadof IP address Hi, I know this can be done but I guess I am not understanding the few notes I have seen on this - can SIP phones be tied to Asterisk using a PC mac address instead of their IP address (obviously I am using DHCP). If someone could please point to the right Wiki or How to I would greatly appreciate it. I would do this by using IP reservations on the DHCP server. Most DHCP servers will allow you to set a reservation of a paricular IP address to a particular MAC address. You may not be able to use this if you have more phones than available IP addresses of course. I couldn't see anything in http://www.voip-info.org/wiki-Asterisk+config+sip.conf that would help your cause directly. Giles ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] xlite regestration fails but calls to thru
Scott, First, you need to get the most recent os for the pix, otherwise you will have a lot of problems with udp packets and translations (including bad checksum on your udp packets). I am running both pix515 and pix501 without a problem with sip and h323. you dont need to open any ports on the pix, because the firewall is an ALG( Application layer gateway). If you have fixup sip enabled on the firewall (there by default), all packets entering port 5060 is examined and rtp ports are open dynamically as needed. The same is true for trusted calls (from inside interface) and untrusted calls (from outside, dmz interfaces). You will need to perform conduit permit commands on the public ip address of Asterisk to allow traffic from untrusted outside interface to come to trusted inside interface on port 5060 with both tcp and udp(all traffic is disabled by default). Please check on the exact syntax of conduit permit with cisco docs. I dont believe you will need to perform this for each RTP port, that should be done automatically by pix ALG. Hope this helps Alex From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Wolfe Sent: Saturday, April 02, 2005 7:03 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] xlite regestration fails but calls to thru While on my network I can register ok with xlite but outside my firewall my Xlite says that regestraion has failed but I am still able to make calls through it. I have opened ports: 5060 udp/tcp and 1-2 udp/tcp is there another port Xlite needs for proper regestration? Is is this a network configuation error on Astrisks part? My Asterisk server is running a IP of 10.0.1.x and my Cisco firewall is passing the public IP address to it from theoutside. Thanks for any advice. -Scott ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Registration to multiple GKs
Charles, I don't think asterisk is a full GK. So if you are asking if asterisk will send out LRQ to the neighbors then I don't believe it would. As far as registering with multiple gk, I wanted to correct myself. An endpoint/gw can register with one primary gk and a number of backup gk. If the primary gk fails, then request will be sent to backup gk in the order of registration. Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Charles Wang Sent: Sunday, April 03, 2005 7:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Registration to multiple GKs Is it possible to run Asterisk with another GKs using Neighbor mode? If it is possible, we can run asterisk with several gnugks. On Apr 2, 2005 10:41 PM, Alex Vishnev [EMAIL PROTECTED] wrote: I don't think you can. The rules of h323 is so that you can register with a single gk at a time. Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of VoIP Newbie Sent: Saturday, April 02, 2005 6:37 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Registration to multiple GKs Hi all, How can I configure chan_h323 or chan_oh323 to register to multiple GK and route calls in-between? Many thanks. Newbie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Charles ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] H.323 call '.....' cleared,reason 8 (Transport failure)
Cenk,
Are you sure that remote will handle H245
tunneling? If the remote does not know how to do that, you will get transport
failure. I would suggest doing FastStart instead and
see if you are getting the same results. Of course, you can verify that the
remote can handle faststart as well.
Alex
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Cenk Yabas
Sent: Saturday, April 02, 2005
6:20 AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] H.323
call '.' cleared,reason 8 (Transport failure)
I
installed the oh323 channel driver and registered to the gate keeper
succesfully.
I come
through the GK, ring the dialed number forabout 0.5 seconds
andloose the line.I contacted the GKand they report that they
receive the correct dialstring to route the call but the call is ended by my
side.
The
dialstring looks like this:
exten
= _.,1,Dial(OH323/${EXTEN},60,r)
I use the
following channel driver:
asterisk-oh323-0.7.1
openh323-Janus_patch4-src
pwlib-Janus_patch4-src
and the
message on asterisk console looks like this:
--
Registered with gatekeeper '[EMAIL PROTECTED]'.
--
Executing Dial(SIP/2000-c9fc, OH323/0012029361212|60|r)
in new stack
-- H.323
call to 0012029361212 with codec(s) g729
-- Called
0012029361212
-- H.323
call 'ip$localhost/2209' cleared, reason 8 (Transport failure)
--
OH323/L2209 is circuit-busy
-- Hungup
'OH323/L2209'
==
Everyone is busy/congested at this time (1:0/1/0)
== Auto
fallthrough, channel 'SIP/2000-c9fc' status is 'CONGESTION'
--
Executing Dial(SIP/2000-c9fc, OH323/h|60|r) in new
stack
-- H.323
call to h with codec(s) g729
-- Called
h
-- Hungup
'OH323/L2210'
== Spawn
extension (local, h, 1) exited non-zero on 'SIP/2000-c9fc'
-- H.323
call 'ip$localhost/2210' cleared, reason 1 (Cleared by local user)
My oh323
configuration:
Configuration
of OpenH323 channel driver
--
Version: 0.7.1
Listening on address: 0.0.0.0:1720
Gatekeeper used: [EMAIL PROTECTED]
(Registered)
FastStart/H245Tunnelling/H245inSetup: OFF/ON/ON
Supported formats in pref. order: g7290
Jitter buffer limits (min/max): 20-100 ms
TCP port range: 5000 - 31000
UDP (RAS) port range: 5000 - 31000
UDP (RTP) port range: 1 - 2
IP Type-of-Service value: 0
User input mode: 2
Max number of inbound H.323 calls: 10
Max number of outbound H.323 calls: 10
Max number of simultaneous H.323 calls: 20
Max call rate (ingress direction): 1.00/30
What might be the problem?
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RE: [Asterisk-Users] Registration to multiple GKs
I don't think you can. The rules of h323 is so that you can register with a single gk at a time. Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of VoIP Newbie Sent: Saturday, April 02, 2005 6:37 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Registration to multiple GKs Hi all, How can I configure chan_h323 or chan_oh323 to register to multiple GK and route calls in-between? Many thanks. Newbie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] problem detecting answer on pri card
I have seen that before when you mismatch the type of pri flavor. For example, if you carrier gives you 4ess and you put 5ess in your config. There are subtle differences in packets. I would check the configuration on your carrier side and * side. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Richard Sent: Saturday, April 02, 2005 1:20 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] problem detecting answer on pri card Hi, I have a digium PRI T1 card connecting to my carrier. However it has problems to detect the answer signal on some numbers. For example, 1-800-225-2525 is KLM airline's reservation line. It should answer right away. But * can't detect it is answered and keeps ringing the ip phone. I put a monitor on the channel, and get the answer messages in the channels. So somehow the line is answered but * doesn't know. I don't have a problem to most numbers. The problem only got my attention after one customer reported it. A debug on the pri shows, Ext: 1 Progress Description: Call is not end-to-end ISDN; further call progress information may be available inband. (1) ] So maybe the inband information is not detected by *? Anyone has the same setup, i.e. PRI to your carrier? Can you please dial the number 1-800-225-2525 and have 'pri debug'? I'd like to compare the results. I am not sure if it is * or just my * configuration. Your help is highly appreciated. I am really stuck here. Thanks, Richard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Codec not negotiating
Clay, It looks like you have the order of the codecs in [general] section as g729, then ulaw. Try reversing them and see if it helps. You may also view the order in the friend section as well. If that works, you may have to setup 2 peers in sip.conf. one for faxing with ulaw, and one with voice with g729. I know thats ugly, but it should work. HTH Alex From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Clay Reiche Sent: Friday, April 01, 2005 3:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Subject: [Asterisk-Users] Codec not negotiating ok... I've trying to fix this for days... I have a sip device that registers with my *. The sip device is ONLY set up to use ulaw. My asterisk server sends ALL PSTN calls to a Sonus gateway/softswitch. When I place a PSTN call, the sip device sends the INVITE with SDP and the ONLY codec option is ulaw. Asterisk then turns around and sends an INVITE with SDP to the Sonus gateway with ulaw as the first option and g729 as a second option. The Sonus sees the TWO options and ALWAYS chooses g729. The codec negotiation fails and the call never completes. I understand that the TWO options are sent because I have no peer set up for the Sonus in my sip.conf and it defaults to the [general] codec settings which are ulaw and g729. However, MOST of my calls to the Sonus ARE using g729, only a few need to use ulaw. (for faxing) So I can't restrict the Sonus peer to only ulaw... Here is my question:(finally...sorry:)) Can I force asterisk to send ONLY my prefered codec?(the first one in the INVITE) or is this only fixed by pleading with the people who run the Sonus sofswitch to stop ignoring my preferred codec? or is there some other solution? Any suggestions would be very appreciated! CONFIG FILES: Sip.Conf: [general] context=default ; Default context for incoming calls ;recordhistory=yes ; Record SIP history by default ; (see sip history / sip no history) ;realm=mydomain.tld ; Realm for digest authentication ; defaults to asterisk ; Realms MUST be globally unique according to RFC 3261 ; Set this to your host name or domain name port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=no ; Enable DNS SRV lookups on outbound calls ; Note: Asterisk only uses the first host ; in SRV records ; Disabling DNS SRV lookups disables the ; ability to place SIP calls based on domain ; names to some other SIP users on the Internet ;pedantic=yes ; Enable slow, pedantic checking for Pingtel ; and multiline formatted headers for strict ; SIP compatibility (defaults to no) ;tos=184 ; Set IP QoS to either a keyword or numeric val ;tos=lowdelay ; lowdelay,throughput,reliability,mincost,none ;maxexpirey=3600 ; Max length of incoming registration we allow ;defaultexpirey=120 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY ;videosupport=yes ; Turn on support for SIP video disallow=all ; First disallow all codecs allow=g729 allow=ulaw ; Allow codecs in order of preference ;allow=alaw ;allow=g723.1 ;allow=ilbc ; Note: codec order is respected only in [general] ;musicclass=default ; Sets the default music on hold class for all SIP calls ; This may also be set for individual users/peers ;language=en ; Default language setting for all users/peers ; This may also be set for individual users/peers ;relaxdtmf=yes ; Relax dtmf handling ;rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity ; when we're not on hold ;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP activity ; when we're on hold (must be rtptimeout) ;trustrpid = no ; If Remote-Party-ID should be trusted ;progressinband=no ; If we should generate in-band ringing always useragent=Abox SS1.0 ; Allows you to change the user agent string ;nat=no ; NAT settings ; yes = Always ignore info and assume NAT ; no = Use NAT mode only according to RFC3581 ; never = Never attempt NAT mode or RFC3581 support ; route = Assume NAT, don't send rport (work around more UNIDEN bugs) ;usereqphone=no [8138644418] type=friend username=8138644418 secret=C34589Y host=dynamic nat=yes context=from-sip callerid=8138644418 canreinvite=yes mailbox=8138644418 accountcode=accxx_group disallow=all allow=g729 allow=ulaw ## extensions.conf: [general] static=yes writeprotect=no [globals] [local] ; ; Master context for local, toll-free, and iaxtel calls only ; include = default include = parkedcalls include = iaxtel700 include = iaxprovider include = from-sip [default] include = from-sip [from-sip] exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED]) exten = 18138644418,4,Dial(IAX2/poseidon:[EMAIL PROTECTED]/[EMAIL
RE: [Asterisk-Users] Problems editing oh323 configuration parameters
You dont have any codecs configured in your oh323 conf. also FastStart with H245 tunneling should be enabled to get the best call-setup out of h323. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cenk Yabas Sent: Thursday, March 31, 2005 7:18 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Problems editing oh323 configuration parameters Checking the oh323 configuration on asterisk console gives the following result below. I'm editing the /etc/asterisk/oh323.conf file to correct the parameters, but the result doesn't change. I didn't receive any error massages during the installation of asterisk-oh323-0.7.1 channel driver. So what might be wrong? localhost*CLI oh323 show conf localhost*CLI Configuration of OpenH323 channel driver -- Version: 0.7.1 Listening on address: :1720 Gatekeeper used: Failed FastStart/H245Tunnelling/H245inSetup: OFF/OFF/OFF Supported formats in pref. order: Jitter buffer limits (min/max): 20-100 ms TCP port range: 5000 - 31000 UDP (RAS) port range: 5000 - 31000 UDP (RTP) port range: 1 - 2 IP Type-of-Service value: 0 User input mode: 2 Max number of inbound H.323 calls: 0 Max number of outbound H.323 calls: 0 Max number of simultaneous H.323 calls: -1 Max call rate (ingress direction): 99.00/30 Thanks in advance for any help, Cenk Yabas. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Installing asterisk and components
Checkout http://www.voip-wiki.org as it relates to asterisk. There are a number of useful guides on how to setup and run asterisk. Btw, all the config files should be located in /etc/asterisk. RH9 should be fine to run asterisk. Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, March 31, 2005 7:29 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Installing asterisk and components In which directories I should install asterisk, chan_capi, and modem driver? And did I forgot something to get asterisk functional? what is best way to test quick is the pbx working, at this point I only have HFC card for external isdn lines? I have RH9 so Linux kernel should be fine? Thank you for your answers This mail sent through L-secure: http://www.l-secure.net/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] cmd Authenticiation
Simon, I am not sure if I understand you question properly. However, you can configure password for each user (peer or friend) in corresponding channel configuration file (i.e. sip.conf) HTH Alex From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Simon Sent: Wednesday, March 30, 2005 10:46 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] cmd Authenticiation Hi folks, Sorry to post a simple command, I am deep into this and hope any help from the experts. I am using the command Authenticate as explained in wi-ki:I am managed to authenticiate with a single global passwordbut my requirement will every user have their own password and contexts to callPlease help meThank youSimon ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Concurrent Call in Asterisk
Stephen, You should be able to setup what you want. For example, asterisk sip peer will register with your provider. The IP/analog phones will attempt outbound calls which will be sent to this provider. What you need to determine is how your provider bills for the calls. If they bill flat, then you can have 1 user sharing the same account. Otherwise, you may want to check with the provider. HTH Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Sent: Thursday, March 31, 2005 8:56 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Concurrent Call in Asterisk Hi All, Is it possible to have only one SIP account that is shared by several users ? I am currently setting up one asterisk box for a small company (around 7 users). Can all of them make simultaneous call using only one SIP account for termination or I have to setup individual account for all of them (which will be very troublesome on my side as I have to keep reminding them to top up , would be good if I just manage one account) ? Thanks in advance. Stephen ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help with Application Development in Asterisk
All, I need some help figuring out the best way to write applications for asterisk. I am trying to implement something similar to astcc pre-paid application where the application will need to play voice prompts, collect tones and perform queries over TCP sockets. It will also need to redirect signaling channel if needed. Looking at astcc (Perl AGI Module), I saw that a new instance of perl was spawned for every call. This is not very scalable. Looking at some alternative I found that there is a manager interface that can monitor channels. However, I am not sure if this is a best approach either. Can someone recommend/comment on their experience writing applications? What method was chosen and why? Sincerely, TIA Alex ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
